Remove unused properties from MediaContentDescription
Bug: webrtc:8620 Change-Id: I0597495a93aa7507177cf7bcbb8c6d35ee90ae0d Reviewed-on: https://webrtc-review.googlesource.com/30500 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21150}
This commit is contained in:
@ -63,8 +63,6 @@ enum {
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MSG_FIRSTPACKETRECEIVED,
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};
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static const int kAgcMinus10db = -10;
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static void SafeSetError(const std::string& message, std::string* error_desc) {
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if (error_desc) {
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*error_desc = message;
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@ -1491,9 +1489,6 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
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AudioSendParameters send_params = last_send_params_;
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RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
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&send_params);
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if (audio->agc_minus_10db()) {
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send_params.options.adjust_agc_delta = kAgcMinus10db;
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}
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bool parameters_applied = media_channel()->SetSendParameters(send_params);
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if (!parameters_applied) {
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@ -469,7 +469,6 @@ static bool AddStreamParams(
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for (size_t i = 0; i < ssrcs.size(); ++i) {
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stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
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}
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content_description->set_multistream(true);
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}
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// Generate extra ssrc for include_flexfec_stream case.
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if (include_flexfec_stream) {
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@ -478,7 +477,6 @@ static bool AddStreamParams(
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std::vector<uint32_t> flexfec_ssrcs;
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GenerateSsrcs(*current_streams, 1, &flexfec_ssrcs);
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stream_param.AddFecFrSsrc(ssrcs[0], flexfec_ssrcs[0]);
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content_description->set_multistream(true);
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} else if (!ssrcs.empty()) {
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RTC_LOG(LS_WARNING)
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<< "Our FlexFEC implementation only supports protecting "
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@ -736,7 +734,6 @@ static bool CreateMediaContentOffer(
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if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
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offer->set_rtcp_reduced_size(true);
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}
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offer->set_multistream(session_options.is_muc);
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offer->set_rtp_header_extensions(rtp_extensions);
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if (!AddStreamParams(sender_options, session_options.rtcp_cname,
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@ -1867,7 +1864,6 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
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crypto_suites, audio_rtp_extensions, current_streams, audio.get())) {
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return false;
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}
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audio->set_lang(lang_);
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bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
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SetMediaProtocol(secure_transport, audio.get());
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@ -124,7 +124,6 @@ struct MediaSessionOptions {
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bool HasMediaDescription(MediaType type) const;
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DataChannelType data_channel_type = DCT_NONE;
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bool is_muc = false;
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bool vad_enabled = true; // When disabled, removes all CN codecs from SDP.
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bool rtcp_mux_enabled = true;
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bool bundle_enabled = false;
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@ -174,11 +173,6 @@ class MediaContentDescription : public ContentDescription {
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cryptos_ = cryptos;
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}
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CryptoType crypto_required() const { return crypto_required_; }
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void set_crypto_required(CryptoType type) {
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crypto_required_ = type;
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}
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const RtpHeaderExtensions& rtp_header_extensions() const {
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return rtp_header_extensions_;
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}
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@ -209,9 +203,6 @@ class MediaContentDescription : public ContentDescription {
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bool rtp_header_extensions_set() const {
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return rtp_header_extensions_set_;
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}
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// True iff the client supports multiple streams.
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void set_multistream(bool multistream) { multistream_ = multistream; }
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bool multistream() const { return multistream_; }
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const StreamParamsVec& streams() const {
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return streams_;
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}
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@ -256,9 +247,6 @@ class MediaContentDescription : public ContentDescription {
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void set_conference_mode(bool enable) { conference_mode_ = enable; }
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bool conference_mode() const { return conference_mode_; }
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void set_partial(bool partial) { partial_ = partial; }
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bool partial() const { return partial_; }
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// https://tools.ietf.org/html/rfc4566#section-5.7
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// May be present at the media or session level of SDP. If present at both
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// levels, the media-level attribute overwrites the session-level one.
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@ -275,13 +263,10 @@ class MediaContentDescription : public ContentDescription {
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int bandwidth_ = kAutoBandwidth;
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std::string protocol_;
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std::vector<CryptoParams> cryptos_;
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CryptoType crypto_required_ = CT_NONE;
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std::vector<webrtc::RtpExtension> rtp_header_extensions_;
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bool rtp_header_extensions_set_ = false;
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bool multistream_ = false;
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StreamParamsVec streams_;
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bool conference_mode_ = false;
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bool partial_ = false;
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webrtc::RtpTransceiverDirection direction_ =
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webrtc::RtpTransceiverDirection::kSendRecv;
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rtc::SocketAddress connection_address_;
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@ -333,27 +318,12 @@ class MediaContentDescriptionImpl : public MediaContentDescription {
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class AudioContentDescription : public MediaContentDescriptionImpl<AudioCodec> {
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public:
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AudioContentDescription() :
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agc_minus_10db_(false) {}
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AudioContentDescription() {}
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virtual ContentDescription* Copy() const {
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return new AudioContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
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const std::string &lang() const { return lang_; }
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void set_lang(const std::string &lang) { lang_ = lang; }
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bool agc_minus_10db() const { return agc_minus_10db_; }
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void set_agc_minus_10db(bool enable) {
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agc_minus_10db_ = enable;
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}
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private:
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bool agc_minus_10db_;
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private:
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std::string lang_;
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};
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class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
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@ -553,7 +523,6 @@ class MediaSessionDescriptionFactory {
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// TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter
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// and setter.
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SecurePolicy secure_ = SEC_DISABLED;
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std::string lang_;
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const TransportDescriptionFactory* transport_desc_factory_;
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};
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@ -1469,42 +1469,6 @@ TEST_F(MediaSessionDescriptionFactoryTest,
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EXPECT_FALSE(dcd->has_ssrcs()); // No StreamParams.
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}
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TEST_F(MediaSessionDescriptionFactoryTest, TestPartial) {
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MediaSessionOptions opts;
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AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
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AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
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f1_.set_secure(SEC_ENABLED);
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std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
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ASSERT_TRUE(offer.get() != NULL);
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const ContentInfo* ac = offer->GetContentByName("audio");
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const ContentInfo* vc = offer->GetContentByName("video");
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const ContentInfo* dc = offer->GetContentByName("data");
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AudioContentDescription* acd = const_cast<AudioContentDescription*>(
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static_cast<const AudioContentDescription*>(ac->description));
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VideoContentDescription* vcd = const_cast<VideoContentDescription*>(
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static_cast<const VideoContentDescription*>(vc->description));
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DataContentDescription* dcd = const_cast<DataContentDescription*>(
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static_cast<const DataContentDescription*>(dc->description));
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EXPECT_FALSE(acd->partial()); // default is false.
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acd->set_partial(true);
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EXPECT_TRUE(acd->partial());
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acd->set_partial(false);
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EXPECT_FALSE(acd->partial());
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EXPECT_FALSE(vcd->partial()); // default is false.
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vcd->set_partial(true);
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EXPECT_TRUE(vcd->partial());
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vcd->set_partial(false);
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EXPECT_FALSE(vcd->partial());
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EXPECT_FALSE(dcd->partial()); // default is false.
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dcd->set_partial(true);
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EXPECT_TRUE(dcd->partial());
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dcd->set_partial(false);
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EXPECT_FALSE(dcd->partial());
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}
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// Create a typical video answer, and ensure it matches what we expect.
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TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswerRtcpMux) {
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MediaSessionOptions offer_opts;
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@ -2412,7 +2376,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
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VideoContentDescription* desc = static_cast<VideoContentDescription*>(
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offer->GetContentDescriptionByName(cricket::CN_VIDEO));
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ASSERT_TRUE(desc != NULL);
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EXPECT_TRUE(desc->multistream());
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const StreamParamsVec& streams = desc->streams();
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// Single stream.
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ASSERT_EQ(1u, streams.size());
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@ -2453,7 +2416,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
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VideoContentDescription* desc = static_cast<VideoContentDescription*>(
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offer->GetContentDescriptionByName(cricket::CN_VIDEO));
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ASSERT_TRUE(desc != nullptr);
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EXPECT_TRUE(desc->multistream());
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const StreamParamsVec& streams = desc->streams();
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// Single stream.
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ASSERT_EQ(1u, streams.size());
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@ -2493,7 +2455,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateNoFlexfecSsrcs) {
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VideoContentDescription* desc = static_cast<VideoContentDescription*>(
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offer->GetContentDescriptionByName(cricket::CN_VIDEO));
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ASSERT_TRUE(desc != nullptr);
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EXPECT_FALSE(desc->multistream());
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const StreamParamsVec& streams = desc->streams();
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// Single stream.
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ASSERT_EQ(1u, streams.size());
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